1. Field of the Invention
The invention relates to apparatus and methods for providing multi-media communication over a private communication network of the type known as a local area network (LAN).
2. Description of the Prior Art
Many types of LANs are known, but they are fundamentally data networks. Local private telephone networks are also well known. However, to date, there has been no effective integration of voice and data technoloay in a private communication system or LAN.
Integration of voice and data has been accomplished on public communication systems, such as those in accordance with the Integrated Services Digital Network (ISDN) specification. But ISDN, like other public telephone systems, is fundamentally circuit switched, i.e., each communication is afforded a dedicated channel for the duration of the communication which is an inherently inefficient system, because typically each subscriber utilizes its allotted channel for a relatively small amount of the available time. Furthermore, communications are limited because there is only a portion of the available bandwidth allotted to each subscriber. Also, switching is centralized so that a failure at the switching center can result in failure of the entire network. These inefficiencies can render it prohibitively expensive to provide the necessary channel capacity in a private network.
This problem can be alleviated in a LAN by the use of token ring technology wherein the entire system bandwidth is made available to each subscriber on a random access basis by providing packet switched rather than circuit switched transfer of information. In such token ring systems the terminals are interconnected in a ring architecture and access to the network is provided by circulating tokens around the ring. Such systems have been used quite effectively for asynchronous applications, such as data transmission, but have not been readily adaptable for synchronous communications, such as voice, which cannot be interrupted and must be received essentially in real time. Heretofore it has not been possible to effectively integrate data and digitized voice on a token ring network so as to afford adequate capacity for data transmission while at the same time effecting real-time voice transmission with no loss of voice information.
Digital telephone networks have been developed which provide a wide variety of useful call processing features. However, most such systems control call processing from a central control station. This adds expense to the system and, furthermore, risks failure of the entire network, as explained above. It is possible to distribute the call processing functions to each telephone device in the network by providing, at each such device, an address table containing all of the addresses of all of the stations in the network, but this also adds significantly to the expense of the system and complicates the addition and deletion of stations.
Several attempts have been made to provide integrated voice, data and video communication on private LANs, but all have required, to some degree, a centralized control. Examples of such systems are disclosed in U.S. Pat. Nos. 4,866,704 and 4,843,606 which provide for integrated voice and data services on a token ring network, but in both systems switching is controlled by an external PBX or PABX unit. Another packet-switched integrated voice and data system is disclosed in U.S. Pat. No. 4,663,758, but it utilizes a network control center which stores a central data base for controlling the system.
U.S. Pat. No. 4,757,497 discloses an integrated voice and data system in which one of a plurality of interconnected nodes is a digital voice switch for the entire network. Furthermore, the system uses frequency division multiplexing.
Another approach is utilized in U.S. Pat. No. 4,557,651, which discloses a system having dual token rings, one of which is a time division multiplex ring for carrying voice, with one of several nodes on the ring acting as a master controller for any given communication.
There is a need for a private communication system which can effectively integrate voice and data communications over a single communication medium, while providing fully distributed control.
One commonly used feature of digital telephone systems is conference calling, wherein more than two parties are simultaneously in conversation with one another. In a packet-switched network, this means that each party to the conference must send its packets of voice information to each of the parties to the conference and must, in turn, receive voice packets from each party to the conference. This significantly multiplies the number of packets which must be handled by the network in any given unit of time. For a two-party communication there are only two packets, one originating at each party. For a three-way conference there are six packets, since each party must send packets to each of the other two parties. In packet-switched networks, such as token ring networks, there are a plurality of stations or node concentrators connected in the ring, with each station being coupled to a plurality of nodes or terminals. Typically, such station apparatus has a limited packet-handling capacity. Thus, the implementation of call conferencing may severely limit the number of nodes which can be coupled to a station and thereby limit the size of the network.